1. Field of the Invention
This invention relates to digital audio decoders that decode digital audio signals (or bit stream data) which are compressed by sub-band coding methods such as MPEG/Audio signals, ATRAC signals and AC-3 signals (where xe2x80x98MPEGxe2x80x99 stands for xe2x80x98Moving Picture Experts Groupxe2x80x99, and xe2x80x98ATRACxe2x80x99 stands for xe2x80x98Adaptive Transform Acoustic Codingxe2x80x99).
2. Description of the Related Art
Conventionally, there are provided various types of compression methods for compressing digital audio signals, one of which is known as the MPEG/Audio standard. FIG. 5 shows an example of a data compression circuit based on the aforementioned standard. Input digital audio signals Da are partitioned into blocks (namely, frames), each of which contains a prescribed number of samples. In the data compression circuit shown in FIG. 4, the input digital audio signals Da are processed by two paths. A first path brings the digital audio signals Da to a filter bank 1 in which they are divided into sub-band signals of thirty-two bands that have equal bandwidths respectively. Each of the sub-band signals is down-sampled to {fraction (1/32)} of the sampling frequency. Then, the sub-band signals are forwarded to a scale factor extraction normalization circuit 2, wherein a sample having a maximal absolute value is detected from each frame of the sub-band signals. The detected value is subjected to quantization to produce a specific value, which is called a scale factor. Using the scale factors, the sub-band signals are subjected to division process and are then subjected to normalization into a prescribed range of values within xc2x11.
A second path brings the digital audio signals Da to an auditory psychology analysis (or auditory perception analysis) block 3 in which frequency spectra are calculated by the fast Fourier transform (FFT). Based on the calculated frequency spectra, the auditory psychology analysis block 3 produces masking thresholds for the sub-band signals respectively, namely allowable quantization noise power. A bit allocation block 4 operates under the restriction of the output of the auditory psychology analysis block 3 and a prescribed number of bits that can be used in one frame, which is determined by the bit rate. Under the aforementioned restriction, the bit allocation block 4 performs repeated loop processes to determine numbers of quantized bits (hereinafter, referred to as xe2x80x98quantization bit numbersxe2x80x99) with respect to sub-bands respectively. Using the quantization bit numbers set for the sub-bands respectively, the quantization block 5 performs quantization on the sub-band signals output from the scale factor extraction normalization circuit 2. That is, the quantization block 5 produces xe2x80x98quantizedxe2x80x99 sub-band samples. A bit stream generation block 6 combines the quantized sub-band samples, bit allocation information and scale factor for each of the sub-bands together in a multiplexing manner. In addition, a header is added to them to create a bit stream, which is output from the bit stream generation block 6.
FIG. 6 shows an example of a configuration of a decoder (or data expansion circuit) that decodes the bit stream, which is produced by the data compression circuit of FIG. 4. Herein, a bit allocation information and scale factor extraction block 11 extracts the bit allocation information and scale factor from the bit stream. In response to the bit allocation information, an inverse quantization circuit 12 reads bit strings respectively corresponding to thirty-two sub-band samples from the bit stream, wherein the bit strings are subjected to inverse quantization with respect to each of the sub-band samples and are then subjected to multiplication by the scale factors. Thus, the inverse quantization circuit 12 produces xe2x80x98inversely quantizedxe2x80x99 sub-band signals, which are synthesized together to reproduce the original digital audio signals by a sub-band synthesis filter bank 13.
Recently, so-called digital sound sources based on the MPEG/Audio standard are widely used in a variety of fields such as pinball game machines, which are widely used in amusement places in Japan. FIG. 6 shows a configuration of a musical tone generation circuit that operates based on the MPEG/Audio standard. Herein, reference numerals 21 designate MPEG/Audio sound sources that contain memories for storing musical tone data, which are made in forms of bit streams respectively, and readout circuits for reading data from the memories respectively. Reference numerals 22 designate decoders (see FIG. 5) that expand output data of the MPEG/Audio sound sources to restore original PCM musical tone data (where xe2x80x98PCMxe2x80x99 stands for xe2x80x98PulseCode Modulationxe2x80x99). Reference numerals 23 designate multipliers that perform gain controls on outputs of the decoders 22. Reference numeral 24 designates an adder that adds together outputs of the multipliers 23. The above describes an example of the configuration of the musical tone generation circuit that is applied to the pinball game machine, for example. This musical tone generation circuit normally provides plural sound sources for multiple channels. That is, the plural sound sources produce MPEG/Audio digital musical tone signals, which are synthesized together to form composite musical tone signals.
The decoder 22 shown in FIG. 6 has processes regarding inverse quantization and sub-band synthesis filter bank, wherein the sub-band synthesis filter bank 13 is configured by a RAM having a relatively large storage capacity. For this reason, the aforementioned musical tone generation circuit of the MPEG/Audio standard, which provides the decoders 22 subsequently to the sound sources 21, bears a problem because the total storage capacity should be increased so much.
It is well known that the conventional digital audio devices use so-called bass boost circuits that amplify low-frequency components of sound. The musical tone generation circuit of the MPEG/Audio standard additionally provides bass boost circuits subsequently to the decoders 22. However, such a configuration causes a problem due to complexity of circuitry because the bass boost circuits should be provided independently of the decoders 22.
In the fields of the digital audio techniques in these days, so-called surround effect techniques are frequently used to enhance richness of sounds. FIG. 8 shows an example of a sound effect circuit, which inputs left-channel signals Li and right-channel signals Ri. Herein, a subtracter 25 produces difference signals between the left-channel signals Li and right-channel signals Ri. A low-pass filter (LPF) filters low frequency components of the difference signals, which are applied to multipliers 26, 27 respectively. The multiplier 26 multiplies them by a positive multiplication coefficient xe2x80x98axe2x80x99, while the multiplier 27 multiplies them by a negative multiplication coefficient xe2x80x98xe2x88x92axe2x80x99. An adder 28 adds together the output of the multiplier 26 and the left-channel signals Li, while an adder 29 adds together the output of the multiplier 27 and the right-channel signals Ri. Thus, the surround effect circuit outputs surround-effect imparted left-channel signals Lo and surround-effect imparted right-channel signals Ro.
It is possible to realize surround effects on musical tone signals of multiple channels. In that case, the musical tone signals are mixed together over the multiple channels with respect to the left channel and right channel respectively. This provides uniform surround effects on all of the channels. However, this is disadvantageous in the prescribe case where one channel is given monaural signals while another channel (left or right channel) is given stereophonic signals because the aforementioned surround effect circuit mistakenly produces mixed signals of two channels as Lo and Ro in FIG. 8.
Conventionally, a variety of configurations and techniques are proposed for processing of digital audio data. For example, Japanese Patent Unexamined Publication No. Hei 8-36399 discloses a processing device in which gain control is made between inverse quantization and quantization of bit streams. Japanese Patent Unexamined Publication No. 2000-29498 discloses a mixing technique using quantization and data reconstruction on compressed digital audio signals of divided frequency bands. Japanese Patent Unexamined Publication No. Hei 9-148940 discloses an improvement in bass boost process on synthesis of compressed data of divided frequency bands. However, none of the aforementioned publications teaches an effective method for solving the aforementioned problems.
It is an object of the invention to provide a digital audio decoder that is reduced in total storage capacity and is simplified in circuit configuration on decoding of compressed digital audio data of divided frequency bands.
It is another object of the invention to provide a digital audio decoder that is capable of imparting desired surround effects on multiple channels independently.
A digital audio decoder of this invention is designed to decode or expand compressed data such as bit stream data, which are compressed based on the MPEG/Audio standard. Herein, inverse quantization circuits perform inverse quantization on plural bit stream data, which are supplied thereto in connection with multiple channels respectively, so that inversely quantized data are produced with respect to a prescribed number (e.g., thirty two) of sub-band samples respectively. The inversely quantized data are combined together among the multiple channels with respect to the prescribed number of the sub-band samples respectively. Then, a filter bank synthesizes together combined data corresponding to all of the sub-band samples, thus reproducing original digital audio signals. Because this invention needs only one filter bank having a relatively large storage capacity, it is possible to reduce the total storage capacity in the digital audio decoder, and it is possible to reduce complexity of circuit configurations in digital audio decoders in manufacture.
In the above, multipliers are provided for use in gain control on the inversely quantized data with respect to the sub-band samples respectively. In addition, it is possible to additionally provide multipliers for amplifying the inversely quantized data of selected sub-band samples corresponding to low-frequency components of sound. This enables bass boost operations to be performed within the decoder.
In addition, it is possible to provide surround effect processing circuits subsequently to the inverse quantization circuits, so desired surround effects are imparted to the inversely quantized data with respect to the sub-band samples respectively. The surround effect processing circuits simply contain multipliers whose coefficients are adequately controlled to achieve selective application of the surround effects among multiple channels.